mirror of http://gerrit.asterisk.org/asterisk
master
20
18
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${ noResults }
19751 Commits (d1fbf03fdfd4238db24a3e8ba7f799edee19863a)
| Author | SHA1 | Message | Date |
|---|---|---|---|
David Vossel
|
d1fbf03fdf |
Fixes chanspy enforced mode lacking a channel_unlock.
(closes issue #19348) Reported by: wdoekes Patches: issue19348_chanspy_missing_channel_unlock.patch uploaded by wdoekes (license 717) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@320506 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
Tilghman Lesher
|
206057bb60 |
Don't crash when the connection fails.
(closes issue #19250) Reported by: seadweller Patches: 20110514__issue19250.diff.txt uploaded by tilghman (license 14) Tested by: seadweller, sum git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@320444 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
Paul Belanger
|
8cd752f1e3 |
Blocked revisions 320393 via svnmerge
........ r320393 | pabelanger | 2011-05-21 01:09:09 -0400 (Sat, 21 May 2011) | 2 lines Solaris compatibility fixes ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@320394 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
David Vossel
|
9764f774c8 |
Fixes issue with ast_tcptls_server_start failing on second attempt to bind.
(closes issue #19289) Reported by: wdoekes Patches: issue19289_delay_old_address_setting_tcptls.patch uploaded by wdoekes (license 717) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@320271 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
Richard Mudgett
|
3e4e3629f3 |
Merged revisions 320235 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r320235 | rmudgett | 2011-05-20 15:38:22 -0500 (Fri, 20 May 2011) | 13 lines The meetme CLI command completion leaves conferences mutex locked. When issuing a meetme kick CLI command and an invalid (non-existent) conference number is specified, pressing Tab leaves the conferences mutex locked and, therefore, all conferences deadlock. Add missing unlock. (closes issue #19336) Reported by: zvision Patches: app_meetme.diff uploaded by zvision (license 798) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@320236 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
Matthew Nicholson
|
9ed4d9d89c |
This commit modifies the way polling is done on TLS sockets.
Because of the buffering the TLS layer does, polling is unreliable. If poll is called while there is data waiting to be read in the TLS layer but not at the network layer, the messaging processing engine will not proceed until something else writes data to the socket, which may not occur. This change modifies the logic around TLS sockets to only poll after a failed read on a non-blocking socket. This way we know that there is no data waiting to be read from the buffering layer. (closes issue #19182) Reported by: st Patches: ssl-poll-fix3.diff uploaded by mnicholson (license 96) Tested by: mnicholson git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@320179 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
David Vossel
|
539b034ed6 |
Blocked revisions 320055 via svnmerge
........ r320055 | dvossel | 2011-05-20 11:38:28 -0500 (Fri, 20 May 2011) | 9 lines chan_sip: Destroy variables on a sip_pvt before copying vars from the sip_peer. (closes issue #19202) Reported by: wdoekes Patches: issue19202_destroy_challenged_invite_chanvars.patch uploaded by wdoekes (license 717) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@320056 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
Terry Wilson
|
0640d5f616 |
Merged revisions 319652 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r319652 | twilson | 2011-05-18 16:04:35 -0700 (Wed, 18 May 2011) | 8 lines Make sure everyone gets an unhold when a transfer succeeds Some phones, like the Snom phones, send a hold to the transfer target after before sending the REFER. We need to make sure that we unhold the parties that are being connected after the masquerade. If Local channels with the /nm option are used when dialing the parties, hold music would still be playing on the transfer target, even after being connected with the transferee. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@319653 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Terry Wilson
|
d0a4962e60 |
Merged revisions 319527 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r319527 | twilson | 2011-05-18 12:56:08 -0700 (Wed, 18 May 2011) | 10 lines Fix app_dial ring groups Revert part of r315643. We need to remove the datastore here as well. The code in bridging code will catch anything that app_dial might miss. (closes issue #19311) Reported by: mspuhler Patches: issue_19311_no_answer.diff uploaded by elguero (license 37) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@319528 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Terry Wilson
|
e59b9085d7 |
Unlink a peer from peers_by_ip when expiring a registration
Review: https://reviewboard.asterisk.org/r/1218/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@319202 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
David Vossel
|
d06dd1262f |
Fixes issue with peer ref-counting during handle_request_subscribe.
(closes issue #19293) Reported by: irroot git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@319144 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Matthew Nicholson
|
229e1541be |
Make sure tcptls_session exists before dereferencing it.
(closes issue #19192) Reported by: stknob Patches: 10-tcptls-unreachable-peer-segfault.patch uploaded by Chainsaw (license 723) Tested by: vois, Chainsaw git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@319141 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Richard Mudgett
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9172658d42 |
Merged revisions 318734 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r318734 | rmudgett | 2011-05-12 20:09:40 -0500 (Thu, 12 May 2011) | 43 lines Merged revisions 318671 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 * The applicable fixes for v1.4 are the SIP deadlock and the in progress masquerade check for multiple parties trying to pickup the same call. issue18654_v1.4.patch uploaded by rmudgett (license 664) * Backported to v1.6.2. issue18654_v1.6.2.patch uploaded by rmudgett (license 664) ........ r318671 | alecdavis | 2011-05-13 10:52:08 +1200 (Fri, 13 May 2011) | 30 lines Fix directed group pickup feature code *8 with pickupsounds enabled Since 1.6.2, the new pickupsound and pickupfailsound in features.conf cause many issues. 1). chan_sip:handle_request_invite() shouldn't be playing out the fail/success audio, as it has 'netlock' locked. 2). dialplan applications for directed_pickups shouldn't beep. 3). feature code for directed pickup should beep on success/failure if configured. Created a sip_pickup() thread to handle the pickup and playout the audio, spawned from handle_request_invite. Moved app_directed:pickup_do() to features:ast_do_pickup(). Functions below, all now use the new ast_do_pickup() app_directed_pickup.c: pickup_by_channel() pickup_by_exten() pickup_by_mark() pickup_by_part() features.c: ast_pickup_call() (closes issue #18654) Reported by: Docent Patches: ast_do_pickup_1.8_trunk.diff.txt uploaded by alecdavis (license 585) Tested by: lmadsen, francesco_r, amilcar, isis242, alecdavis, irroot, rymkus, loloski, rmudgett Review: https://reviewboard.asterisk.org/r/1185/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@318735 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Richard Mudgett
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d12154f767 |
Merged revision 222981 from
https://origsvn.digium.com/svn/asterisk/branches/1.8 Similar deadlock possible when running the Pickup application internally. ------------------------------------------------------------------------ r222981 | dvossel | 2009-10-08 17:04:41 -0500 (Thu, 08 Oct 2009) | 13 lines Deadlock between ast_cel_report_event and ast_do_masquerade chan_sip calls pbx_exec on a pvt's owner channel while only the pvt lock is held. Since pbx_exec calls ast_cel_report_event which attempts to lock the channel, invalid locking order occurs. Channels should be locked before pvt's. (closes issue #15512) Reported by: lmsteffan Patches: ast_cel_deadlock_15512.diff uploaded by dvossel (license 671) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@318636 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Terry Wilson
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fe41f00a47 |
Clean up several chan_sip reference leaks
Several situations in the code could lead to peers or sip_pvt references being leaked. This would cause RTP ports to never be destroyed (leading to exhaustion of all available RTP ports) and memory leaks. The original patch for this issue from rgagnon was the result of an obscene amount of testing and hard work, for which I am very grateful. I did some cleanup and added a few additional refcount fixes that I found. (closes issue #17255) Reported by: kvveltho Patches: tag-1.6.2.17-r309252-sip-dos-mem-leak-fix.diff uploaded by rgagnon (license 1202) Tested by: rgagnon, twilson, wdoekes, loloski Review: https://reviewboard.asterisk.org/r/1101/ Review: https://reviewboard.asterisk.org/r/1207/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@318548 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Terry Wilson
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12e2f9cdeb |
Don't offer video to directmedia callee unless caller offered it as well
Make sure that when directmedia is enabled, that video is not offered to the callee even if it supports it. p->vrtp will not exist since the caller didn't offer video. (closes issue #19195) Reported by: one47 Patches: sip_cant_add_video_rtp uploaded by one47 (license 23) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@318331 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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David Vossel
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d693f81ab6 |
Fixes cases where sip_set_rtp_peer can return too early during media path reset.
(closes issue #19225) Reported by: one47 Patches: sip_set_rtp_peer.patch uploaded by one47 (license 23) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@318230 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Matthew Nicholson
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7eca34890a |
pbx_lua autoservice fixes
Don't start an autoservice in pbx_lua if pbx_lua already started one and don't stop one if we didn't start one. Also start and stop the autoservice when transferring control from and to the pbx. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@317859 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Richard Mudgett
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5081b488fe |
Merged revisions 317719 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r317719 | rmudgett | 2011-05-06 12:59:05 -0500 (Fri, 06 May 2011) | 11 lines Regression after r297603 (Improve handling of REGISTER requests with multiple contact headers.) Uninitialized variable. (issue #18640) (closes issue #18785) Reported by: pnlarsson Patches: issue18785_enegaard.patch uploaded by enegaard (license 1197) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@317720 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Matthew Nicholson
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c4ed0bac68 |
Add a datastore fixup to fix a pbx_lua crash.
(closes issue #19055) Reported by: jamhed Patches: lua_datastore_fixup1.diff uploaded by mnicholson (license 96) Tested by: mnicholson, jamhed git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@317666 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Terry Wilson
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30e8e7dedb |
Merged revisions 317574 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r317574 | twilson | 2011-05-06 00:55:21 -0700 (Fri, 06 May 2011) | 6 lines Re-fix queue round-robin This part of the change for r315596 was incorrect. No bridge occurs when doing a roundrobin dial and no one answers, so this code shouldn't have been removed. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@317575 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
Russell Bryant
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5fa33b9e28 |
Merged revisions 317211 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r317211 | russell | 2011-05-05 13:20:29 -0500 (Thu, 05 May 2011) | 15 lines chan_sip: fix broken realtime peer count, fix memory leak This patch addresses two bugs in chan_sip: 1) The count of realtime peers and users was off. The increment checked the value of the caching option, while the decrement did not. 2) Add a missing regfree() for a regex. (closes issue #19108) Reported by: vrban Patches: missing_regfree.patch uploaded by vrban (license 756) sip_object_counter.patch uploaded by vrban (license 756) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@317255 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Matthew Nicholson
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b9bf3d604e |
Set SO_KEEPALIVE on SIP TCP sockets so that they eventually go away when a peer
abruptly disappears. This mostly occurs after a successful registration. (closes issue #17544) Reported by: marcelloceschia Patches: (modified) tcptls.patch uploaded by st (license 907) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@317195 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
Leif Madsen
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5731697675 |
Merged revisions 317102 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r317102 | lmadsen | 2011-05-05 10:54:46 -0400 (Thu, 05 May 2011) | 8 lines Disable console colourization inside safe_asterisk checks. (closes issue #19213) Reported by: lefoyer Patches: issue19213_strip_color_in_safe_asterisk-svn.patch uploaded by wdoekes (license 717) Tested by: wdoekes, lefoyer ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@317103 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
Sean Bright
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ad0b87f72b |
Merged revisions 316707 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r316707 | seanbright | 2011-05-04 12:08:50 -0400 (Wed, 04 May 2011) | 8 lines If sox fails when processing a voicemail, don't delete the original file. (closes issue #18111) Reported by: sysreq Patches: issue18111_trunk.patch uploaded by seanbright (license 71) Tested by: seanbright ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@316708 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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David Vossel
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1ef5452cd8 |
Fixes one-way-audio when chanspy activated with the 'o' option
(closes issue #18382) Reported by: jkister Patches: 0001-Bugfix-18382-one-way-audio-when-chanspy-activated.patch.txt uploaded by malin (license ) Tested by: firstsip, Greenlightcrm, malin, wdoekes, boroda, dvossel git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@316644 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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David Vossel
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6b94fbacfa |
Fixes session-timers=refuse not being enforced for *caller*
During handle_request_invite, the session timer mode was retrieved from a cached variable. This patch forces a peer lookup of the session timer mode in the case of an incoming invite. (closes issue #18804) Reported by: wdoekes Patches: issue18804_session_timer_refuse_caller.patch uploaded by wdoekes (license 717) issue_18804_v2.diff uploaded by dvossel (license 671) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@316616 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Sean Bright
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1e2ce928a5 |
Honor the C option to MeetMe when L is passed.
This fixes a case that r304773 and friends missed. (closes issue #17317) Reported by: var Patches: meetme-continue-on-l_16218.diff uploaded by var (license 1227) Tested by: seanbright git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@316475 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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David Vossel
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abf3b8cf58 |
Merged revisions 316328 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r316328 | dvossel | 2011-05-03 16:27:59 -0500 (Tue, 03 May 2011) | 10 lines Fixes chan_local crashs in local_fixup() Thanks OEJ for tracking down the issue and submitting the patch. (closes issue #19053) Reported by: oej Tested by: oej Review: https://reviewboard.asterisk.org/r/1158/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@316329 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Tilghman Lesher
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2b32b45889 |
More possible crashes based upon invalid inputs.
(closes issue #18161) Reported by: wdoekes Patches: 20110301__issue18161.diff.txt uploaded by tilghman (license 14) Tested by: wdoekes git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@316093 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Matthew Nicholson
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96ebbd8ddb |
Merged revisions 315891 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r315891 | mnicholson | 2011-04-27 13:57:56 -0500 (Wed, 27 Apr 2011) | 14 lines Fix our compliance with RFC 3261 section 18.2.2. This change optimizes the free_via() function and removes some redundant null checking. It also fixes compliance with RFC 3261 section 18.2.2 by always using the port specified in the Via header for routing responses (even when maddr is not set). Also the htons() function is now used when setting the port. Additional documentation comments have been added in various places to make the logic in the code clearer. (closes issue #18951) Reported by: jmls Patches: issue18951_set_proper_port_from_via.patch uploaded by wdoekes (license 717) (modified) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@315893 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Terry Wilson
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6d933f5b74 |
Merged revisions 315671 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r315671 | twilson | 2011-04-26 15:47:56 -0700 (Tue, 26 Apr 2011) | 11 lines Make sure unregistering a peer unlinks it from the peer container Instead of mostly copying the code from expire_register, just use the function that "does the right thing". (closes issue #16033) Reported by: kkm Patches: 016033-tilgman-fixed-refcount.diff uploaded by kkm (license 888) Tested by: kkm, tilghman, twilson ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@315672 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Terry Wilson
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445ed7b6ee |
Merged revisions 315596 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r315596 | twilson | 2011-04-26 14:16:10 -0700 (Tue, 26 Apr 2011) | 18 lines Allow transfer loops without allowing forwarding loops We try to avoid the situation where two phones may be forwarded to each other causing an infinite loop by storing each dialed interface in a channel datastore and checking the list before dialing out. This works, but currently breaks situations like A calls B, A transfers B to C, B transfers C to A, and A transfers C to B. Since human interaction is happening here and not an automated forwarding loop, it should be allowed. This patch removes the dialed_interfaces datastore when a call is bridged (a suggestion from the brilliant mmichelson). If a call is being bridged, it should be safe to assume that we aren't stuck in a loop. Since we are now handling this is the bridge code, the previous attempts at handling it in app_dial and app_queue are removed. Review: https://reviewboard.asterisk.org/r/1195/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@315643 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Tilghman Lesher
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93351f0d0b |
Merged revisions 315501 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r315501 | tilghman | 2011-04-26 14:18:46 -0500 (Tue, 26 Apr 2011) | 14 lines Fix the bounds-checking code. The code that set the bit within the select bitfield was correct, but the bounds-checking code was not. The change to that line uses the new _bitsize macro for clarity. Also, FD_ZERO macro did not zero-out anything but the first word of the bitfield, so this could have caused problems with modules using that macro with the expanded bitfield. (closes issue #18773) Reported by: jamicque Patches: 20110423__issue18773.diff.txt uploaded by tilghman (license 14) Tested by: chris-mac ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@315502 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Paul Belanger
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781d279901 |
Add back CLI command 'dialplan save'
(closes issue #19140) Reported by: lmadsen Patches: __20110419_dialplan_save.patch.txt uploaded by lmadsen (license 10) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@315393 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Russell Bryant
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fd7ebbc0f9 |
Merged revisions 315257 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r315257 | russell | 2011-04-25 14:28:41 -0500 (Mon, 25 Apr 2011) | 10 lines Be more flexible with unknown chunks in wav files. This patch makes format_wav ignore unknown chunks instead of erroring out on them. (closes issue #18306) Reported by: jhirsch Patches: wav_skip_unknown_blocks.diff uploaded by jhirsch (license 1156) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@315258 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Russell Bryant
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8ac2b5b078 |
Don't link non-cached realtime peers into the peers_by_ip container.
(closes issue #18924) Reported by: wdoekes Patches: issue18924_uncached_realtime_peers_leak-1.6.2.17.patch uploaded by wdoekes (license 717) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@315212 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
Alec L Davis
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1bc9545d2e |
Merged revisions 315051 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r315051 | alecdavis | 2011-04-25 19:06:29 +1200 (Mon, 25 Apr 2011) | 11 lines chan_local:check_bridge() misplaced misplaced ast_mutex_unlock if !p->chan->_bridge->_softhangup path isn't followed, brigde remains locked. (closes issue #19176) Reported by: alecdavis Patches: bug19176.diff.txt uploaded by alecdavis (license 585) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@315052 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Matthew Nicholson
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826448948f |
Merged revisions 311203,314908 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r311203 | mnicholson | 2011-03-17 14:14:37 -0500 (Thu, 17 Mar 2011) | 4 lines Don't hold the pvt lock while streaming a file. ABE-2756 ........ r314908 | mnicholson | 2011-04-22 15:01:48 -0500 (Fri, 22 Apr 2011) | 4 lines Prevent the login thread and the app threads from using the asterisk channel at the same time. ABE-2756 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@314958 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Russell Bryant
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e26c0cb085 |
Recorded merge of revisions 314822 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r314822 | russell | 2011-04-22 09:34:23 -0500 (Fri, 22 Apr 2011) | 11 lines Initialize buffers in getvar and getvarfull. Initialize the buffers used to hold the result from GET VARIABLE or GET VARIABLE FULL. The bug report shows func_read returning garbage in the result. It assumed that the buffer passed in was initialized, like many other functions do. In the more common code path (through the dialplan), it is initialized, so just initialize it here too. (closes issue #19050) Reported by: johnz ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@314823 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Russell Bryant
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7f6b790054 |
Initialize buffers in getvar and getvarfull.
Initialize the buffers used to hold the result from GET VARIABLE or GET VARIABLE FULL. The bug report shows func_read returning garbage in the result. It assumed that the buffer passed in was initialized, like many other functions do. In the more common code path (through the dialplan), it is initialized, so just initialize it here too. (closes issue #19050) Reported by: johnz git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@314778 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Russell Bryant
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b82f20cae0 |
Fix handling of some call parking config options.
This patch adjusts the handling of some call parking config options to fix some issues that have already been addressed in 1.8 and trunk. (closes issue #19167) Reported by: bluecrow76 Patches: asterisk-1.6.2.17.2-fix-build-parkinglot-parked-AST_FEATURE_FLAGS.diff uploaded by bluecrow76 (license 270) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@314776 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Matthew Nicholson
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2939c487cd |
Merged revisions 314607 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r314607 | mnicholson | 2011-04-21 13:19:21 -0500 (Thu, 21 Apr 2011) | 14 lines Added limits to the number of unauthenticated sessions TCP based protocols are allowed to have open simultaneously. Also added timeouts for unauthenticated sessions where it made sense to do so. Unrelated, the manager interface now properly checks if the user has the "system" privilege before executing shell commands via the Originate action. AST-2011-005 AST-2011-006 (closes issue #18787) Reported by: kobaz (related to issue #18996) Reported by: tzafrir ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@314620 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Terry Wilson
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dd044c85d7 |
Don't allocate more space than necessary for a sip_pkt
This extra allocation is a hold-over from when pkt->data was a character array. Now that it is an allocated string, just allocate enough for the sip_pkt. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@314549 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Leif Madsen
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0dc9bdf3ec |
Remove duplicate documentation from func_channel.c
(closes issue #18970) Reported by: IgorG Patches: func_channel.c.doc.diff uploaded by IgorG (license 20) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@314205 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Leif Madsen
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370816bdac |
Update seconds to milliseconds in ast_verb output.
(closes issue #19084) Reported by: smurfix Patches: app_dial.patch uploaded by smurfix (license 547) Tested by: lmadsen, smurfix git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@314202 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
Jonathan Rose
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4751ddd44e |
Fix a Tab Completion bug that occurs due to multiple matches on a substring.
Makes word_match function in cli.c repeat a search for a command string until a proper match is found or the string is searched to the last point. (closes issue #17494) Reported by: ffossard Review: https://reviewboard.asterisk.org/r/1180/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@313859 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Richard Mudgett
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b9c89feadc |
Merged revisions 313545 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r313545 | rmudgett | 2011-04-13 11:21:24 -0500 (Wed, 13 Apr 2011) | 41 lines Asterisk does not hangup a channel after endpoint hangs up. If the call that the dialplan started an AGI script for is hungup while the AGI script is in the middle of a command then the AGI script is not notified of the hangup. There are many AGI Exec commands that this can happen with. The reported applications have been: Background, Wait, Read, and Dial. Also the AGI Get Data command. * Don't wait on the Asterisk channel after it has hung up. The channel is likely to never need servicing again. * Restored the AGI script's ability to return the AGI_RESULT_HANGUP value in run_agi(). It previously only could return AGI_RESULT_SUCCESS or AGI_RESULT_FAILURE after the DeadAGI and AGI applications were merged. (closes issue #17954) Reported by: mn3250 Patches: issue17954_v1.8.patch uploaded by rmudgett (license 664) issue17954_v1.6.2.patch uploaded by rmudgett (license 664) issue17954_v1.4.patch uploaded by rmudgett (license 664) Tested by: rmudgett JIRA SWP-2171 (closes issue #18492) Reported by: devmod Tested by: rmudgett JIRA SWP-2761 (closes issue #18935) Reported by: nvitaly Tested by: astmiv, rmudgett JIRA SWP-3216 (closes issue #17393) Reported by: siby Tested by: rmudgett JIRA SWP-2727 Review: https://reviewboard.asterisk.org/r/1165/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@313579 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Jonathan Rose
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f33df943bc |
fixing stupid mistake with putting code before variable declaration
........ Merged revisions 313433 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r313432 | jrose | 2011-04-12 13:12:29 -0500 (Tue, 12 Apr 2011) | 14 lines reload Chan_dahdi memory leak caused by variables chan_dahdi reloading with variables set via setvar in chan_dahdi.conf would stay in the dahdi_pvt structs for individual channels (causing them to just continue adding the new ones to the list) and also there was a memory leak causes by the conf objects. This patch resolves both of these by using ast_variables_destroy during the loading process. (closes issue #17450) Reported by: nahuelgreco Patches: patch.diff uploaded by jrose (license 1225) Tested by: tilghman, jrose Review: https://reviewboard.asterisk.org/r/1170/ ........ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@313435 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Jonathan Rose
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a9c87b24e4 |
white space change
........ reload Chan_dahdi memory leak caused by variables chan_dahdi reloading with variables set via setvar in chan_dahdi.conf would stay in the dahdi_pvt structs for individual channels (causing them to just continue adding the new ones to the list) and also there was a memory leak causes by the conf objects. This patch resolves both of these by using ast_variables_destroy during the loading process. (closes issue #17450) Reported by: nahuelgreco Patches: patch.diff uploaded by jrose (license 1225) Tested by: tilghman, jrose Review: https://reviewboard.asterisk.org/r/1170/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@313433 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |